Rewrote the hold and netw-before-streaming tests.
Rewrote the hold test. Abstracted out resource handling and created a new fixture for starting and stopping playing. Rewrote network-before-streaming. BUG= TEST=voe_auto_test Review URL: http://webrtc-codereview.appspot.com/331001 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1228 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
parent
398af2337b
commit
188fc35e07
@ -7,12 +7,11 @@
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "gtest/gtest.h"
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int RunInAutomatedMode(int argc, char** argv) {
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// Initialize the testing framework.
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testing::InitGoogleTest(&argc, argv);
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// Run tests.
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return RUN_ALL_TESTS();
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}
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29
src/voice_engine/main/test/auto_test/resource_manager.cc
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29
src/voice_engine/main/test/auto_test/resource_manager.cc
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@ -0,0 +1,29 @@
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "resource_manager.h"
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#include "testsupport/fileutils.h"
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ResourceManager::ResourceManager() {
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std::string filename = "audio_long16.pcm";
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#if defined(WEBRTC_ANDROID)
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long_audio_file_path_ = "/sdcard/" + filename;
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#else
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std::string resource_path = webrtc::test::ProjectRootPath();
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if (resource_path == webrtc::test::kCannotFindProjectRootDir) {
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long_audio_file_path_ = "";
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} else {
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long_audio_file_path_ =
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resource_path + "test/data/voice_engine/" + filename;
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}
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#endif
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}
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30
src/voice_engine/main/test/auto_test/resource_manager.h
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30
src/voice_engine/main/test/auto_test/resource_manager.h
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@ -0,0 +1,30 @@
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_RESOURCE_MANAGER_H_
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#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_RESOURCE_MANAGER_H_
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#include <string>
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class ResourceManager {
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public:
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ResourceManager();
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// Returns the full path to a long audio file.
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// Returns the empty string on failure.
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const std::string& long_audio_file_path() const {
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return long_audio_file_path_;
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}
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private:
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std::string long_audio_file_path_;
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};
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#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_RESOURCE_MANAGER_H_
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@ -18,7 +18,9 @@ class TestErrorObserver;
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// This fixture initializes the voice engine in addition to the work
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// done by the before-initialization fixture. It also registers an error
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// observer which will fail tests on error callbacks.
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// observer which will fail tests on error callbacks. This fixture is
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// useful to tests that want to run before we have started any form of
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// streaming through the voice engine.
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class AfterInitializationFixture : public BeforeInitializationFixture {
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public:
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AfterInitializationFixture();
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@ -0,0 +1,66 @@
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "after_streaming_fixture.h"
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#include <cstring>
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#include "voice_engine_defines.h"
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static const char* kLoopbackIp = "127.0.0.1";
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AfterStreamingFixture::AfterStreamingFixture()
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: channel_(voe_base_->CreateChannel()) {
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EXPECT_GE(channel_, 0);
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const std::string& input_file = resource_manager_.long_audio_file_path();
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ASSERT_FALSE(input_file.empty());
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SetUpLocalPlayback();
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StartPlaying(input_file);
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}
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AfterStreamingFixture::~AfterStreamingFixture() {
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voe_file_->StopPlayingFileAsMicrophone(channel_);
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voe_base_->StopSend(channel_);
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voe_base_->StopPlayout(channel_);
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voe_base_->StopReceive(channel_);
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voe_base_->DeleteChannel(channel_);
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}
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void AfterStreamingFixture::SetUpLocalPlayback() {
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EXPECT_EQ(0, voe_base_->SetSendDestination(channel_, 8000, kLoopbackIp));
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EXPECT_EQ(0, voe_base_->SetLocalReceiver(0, 8000));
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webrtc::CodecInst codec;
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codec.channels = 1;
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codec.pacsize = 160;
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codec.plfreq = 8000;
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codec.pltype = 0;
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codec.rate = 64000;
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strcpy(codec.plname, "PCMU");
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voe_codec_->SetSendCodec(channel_, codec);
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}
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void AfterStreamingFixture::StartPlaying(const std::string& input_file) {
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EXPECT_EQ(0, voe_base_->StartReceive(0));
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EXPECT_EQ(0, voe_base_->StartPlayout(0));
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EXPECT_EQ(0, voe_base_->StartSend(0));
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EXPECT_EQ(0, voe_file_->StartPlayingFileAsMicrophone(
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0, input_file.c_str(), true, true));
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}
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void AfterStreamingFixture::Sleep(long milliseconds) {
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// Implementation note: This method is used to reduce usage of the macro and
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// avoid ugly errors in Eclipse (its parser can't deal with the sleep macro).
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SLEEP(milliseconds);
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}
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@ -0,0 +1,38 @@
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_AFTER_STREAMING_H_
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#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_AFTER_STREAMING_H_
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#include "after_initialization_fixture.h"
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#include "resource_manager.h"
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// This fixture will, in addition to the work done by its superclasses,
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// create a channel and start playing a file through the fake microphone
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// to simulate microphone input. The purpose is to make it convenient
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// to write tests that require microphone input.
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class AfterStreamingFixture : public AfterInitializationFixture {
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public:
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AfterStreamingFixture();
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virtual ~AfterStreamingFixture();
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protected:
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int channel_;
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ResourceManager resource_manager_;
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// Use this sleep function to sleep in test (avoid sleep macro).
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void Sleep(long milliseconds);
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private:
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void SetUpLocalPlayback();
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void StartPlaying(const std::string& input_file);
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};
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#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_AFTER_STREAMING_H_
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@ -0,0 +1,43 @@
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "after_streaming_fixture.h"
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// Note: This class includes sleeps and requires manual verification.
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class ManualHoldTest : public AfterStreamingFixture {
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};
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TEST_F(ManualHoldTest, SetOnHoldStatusBlockAudio) {
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TEST_LOG("Channel not on hold => should hear audio.\n");
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Sleep(2000);
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TEST_LOG("Put channel on hold => should *not* hear audio.\n");
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EXPECT_EQ(0, voe_base_->SetOnHoldStatus(channel_, true));
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Sleep(2000);
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TEST_LOG("Remove on hold => should hear audio again.\n");
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EXPECT_EQ(0, voe_base_->SetOnHoldStatus(channel_, false));
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Sleep(2000);
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TEST_LOG("Put sending on hold => should *not* hear audio.\n");
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EXPECT_EQ(0, voe_base_->SetOnHoldStatus(channel_, true, webrtc::kHoldSendOnly));
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Sleep(2000);
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}
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TEST_F(ManualHoldTest, SetOnHoldStatusBlocksLocalFileAudio) {
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TEST_LOG("Start playing a file locally => "
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"you should now hear this file being played out.\n");
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voe_file_->StopPlayingFileAsMicrophone(channel_);
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EXPECT_EQ(0, voe_file_->StartPlayingFileLocally(
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channel_, resource_manager_.long_audio_file_path().c_str(), true));
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Sleep(2000);
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TEST_LOG("Put playing on hold => should *not* hear audio.\n");
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EXPECT_EQ(0, voe_base_->SetOnHoldStatus(
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channel_, true, webrtc::kHoldPlayOnly));
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Sleep(2000);
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}
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@ -0,0 +1,55 @@
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "after_initialization_fixture.h"
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class NetworkBeforeStreamingTest : public AfterInitializationFixture {
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protected:
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void SetUp() {
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channel_ = voe_base_->CreateChannel();
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}
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void TearDown() {
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voe_base_->DeleteChannel(channel_);
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}
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int channel_;
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};
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TEST_F(NetworkBeforeStreamingTest,
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GetSourceInfoReturnsEmptyValuesForUnconfiguredChannel) {
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char src_ip[32] = "0.0.0.0";
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int src_rtp_port = 1234;
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int src_rtcp_port = 1235;
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EXPECT_EQ(0, voe_network_->GetSourceInfo(
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channel_, src_rtp_port, src_rtcp_port, src_ip));
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EXPECT_EQ(0, src_rtp_port);
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EXPECT_EQ(0, src_rtcp_port);
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EXPECT_STRCASEEQ("", src_ip);
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}
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TEST_F(NetworkBeforeStreamingTest,
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GetSourceFilterReturnsEmptyValuesForUnconfiguredChannel) {
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int filter_port = -1;
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int filter_port_rtcp = -1;
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char filter_ip[32] = "0.0.0.0";
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EXPECT_EQ(0, voe_network_->GetSourceFilter(
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channel_, filter_port, filter_port_rtcp, filter_ip));
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EXPECT_EQ(0, filter_port);
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EXPECT_EQ(0, filter_port_rtcp);
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EXPECT_STRCASEEQ("", filter_ip);
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}
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TEST_F(NetworkBeforeStreamingTest, SetSourceFilterSucceeds) {
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EXPECT_EQ(0, voe_network_->SetSourceFilter(channel_, 0));
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}
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@ -34,7 +34,6 @@
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#include "critical_section_wrapper.h"
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#include "event_wrapper.h"
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#include "thread_wrapper.h"
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#include "testsupport/fileutils.h"
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#ifdef _TEST_NETEQ_STATS_
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#include "../../interface/voe_neteq_stats.h" // Not available in delivery folder
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@ -621,9 +620,7 @@ VoETestManager::VoETestManager()
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voe_rtp_rtcp_(0),
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voe_vsync_(0),
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voe_volume_control_(0),
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voe_apm_(0),
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resource_path_(),
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audio_filename_()
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voe_apm_(0)
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{
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}
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@ -642,18 +639,6 @@ bool VoETestManager::Init() {
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return false;
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}
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#if defined(WEBRTC_ANDROID)
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resource_path_ = "/sdcard/";
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#else
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resource_path_ = webrtc::test::ProjectRootPath();
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if (resource_path_ == webrtc::test::kCannotFindProjectRootDir) {
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TEST_LOG("Failed to get project root directory\n");
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return false;
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}
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resource_path_ += "test/data/voice_engine/";
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#endif
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audio_filename_ = resource_path_ + "audio_long16.pcm";
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voice_engine_ = VoiceEngine::Create();
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if (!voice_engine_) {
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TEST_LOG("Failed to create VoiceEngine\n");
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@ -921,46 +906,6 @@ int VoETestManager::SetUp() {
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return 0;
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}
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int VoETestManager::TestNetworkBeforeStreaming() {
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///////////////////////////////////////////////
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// Network (test before streaming is activated)
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#ifdef _TEST_NETWORK_
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TEST_LOG("\n\n+++ Network tests +++\n\n");
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#ifndef WEBRTC_EXTERNAL_TRANSPORT
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int filter_port = -1;
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int filter_port_rtcp = -1;
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char src_ip[32] = "0.0.0.0";
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char filter_ip[32] = "0.0.0.0";
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TEST_LOG("GetSourceInfo \n");
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int src_rtp_port = 1234;
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int src_rtcp_port = 1235;
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TEST_MUSTPASS(voe_network_->GetSourceInfo(0, src_rtp_port, src_rtcp_port,
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src_ip));
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TEST_MUSTPASS(0 != src_rtp_port);
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TEST_MUSTPASS(0 != src_rtcp_port);
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TEST_MUSTPASS(_stricmp(src_ip, ""));
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TEST_LOG("GetSourceFilter \n");
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TEST_MUSTPASS(voe_network_->GetSourceFilter(0, filter_port, filter_port_rtcp,
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filter_ip));
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TEST_MUSTPASS(0 != filter_port);
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TEST_MUSTPASS(0 != filter_port_rtcp);
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TEST_MUSTPASS(_stricmp(filter_ip, ""));
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TEST_LOG("SetSourceFilter \n");
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TEST_MUSTPASS(voe_network_->SetSourceFilter(0, src_rtp_port));
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#else
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TEST_LOG("Skipping network tests - WEBRTC_EXTERNAL_TRANSPORT is defined \n");
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#endif // #ifndef WEBRTC_EXTERNAL_TRANSPORT
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#else
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TEST_LOG("\n\n+++ Network tests NOT ENABLED +++\n");
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#endif
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return 0;
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}
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int VoETestManager::TestStartStreaming(FakeExternalTransport& channel0_transport) {
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TEST_LOG("\n\n+++ Starting streaming +++\n\n");
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@ -1026,35 +971,8 @@ int VoETestManager::TestStartPlaying() {
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return 0;
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}
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int VoETestManager::TestHoldAndNetEq() {
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int VoETestManager::TestNetEq() {
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#ifdef _TEST_BASE_
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TEST_LOG("Put channel on hold => should *not* hear audio \n");
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// HOLD_SEND_AND_PLAY is the default mode.
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TEST_MUSTPASS(voe_base_->SetOnHoldStatus(0, true));
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SLEEP(2000);
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TEST_LOG("Remove on hold => should hear audio again \n");
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TEST_MUSTPASS(voe_base_->SetOnHoldStatus(0, false));
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SLEEP(2000);
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TEST_LOG("Put sending on hold => should *not* hear audio \n");
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TEST_MUSTPASS(voe_base_->SetOnHoldStatus(0, true, kHoldSendOnly));
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SLEEP(2000);
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if (voe_file_) {
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TEST_LOG("Start playing a file locally => "
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"you should now hear this file being played out \n");
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TEST_MUSTPASS(voe_file_->StartPlayingFileLocally(0, AudioFilename(),
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true));
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SLEEP(2000);
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}
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TEST_LOG("Put playing on hold => should *not* hear audio \n");
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TEST_MUSTPASS(voe_base_->SetOnHoldStatus(0, true, kHoldPlayOnly));
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SLEEP(2000);
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TEST_LOG("Remove on hold => should hear audio again \n");
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if (voe_file_) {
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TEST_MUSTPASS(voe_file_->StopPlayingFileLocally(0));
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}
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TEST_MUSTPASS(voe_base_->SetOnHoldStatus(0, false));
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SLEEP(2000);
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NetEqModes mode;
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TEST_MUSTPASS(voe_base_->GetNetEQPlayoutMode(0, mode));
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TEST_MUSTPASS(mode != kNetEqDefault);
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@ -1412,23 +1330,24 @@ int VoETestManager::TestCodecs() {
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}
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int VoETestManager::DoStandardTest() {
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// Ensure we have all input files:
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TEST_MUSTPASS(!strcmp("", AudioFilename()));
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TEST_LOG("\n\n+++ Base tests +++\n\n");
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if (SetUp() != 0) return -1;
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if (TestNetworkBeforeStreaming() != 0) return -1;
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// TODO(qhogpat): this gets verified way later - quite ugly. Make sure to
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// put this into setup when rewriting the test that requires this.
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TEST_MUSTPASS(voe_rtp_rtcp_->SetRTCP_CNAME(0, "Niklas"));
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TEST_MUSTPASS(voe_rtp_rtcp_->SetLocalSSRC(0, 1234));
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voe_network_->SetSourceFilter(0, 0);
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FakeExternalTransport channel0_transport(voe_network_);
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if (TestStartStreaming(channel0_transport) != 0) return -1;
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if (TestStartPlaying() != 0) return -1;
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if (TestHoldAndNetEq() != 0) return -1;
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if (TestNetEq() != 0) return -1;
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if (TestCodecs() != 0) return -1;
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/////////////////////////
|
||||
|
@ -14,15 +14,15 @@
|
||||
#include <stdio.h>
|
||||
#include <string>
|
||||
|
||||
#include "resource_manager.h"
|
||||
#include "voe_audio_processing.h"
|
||||
#include "voe_base.h"
|
||||
#include "voe_dtmf.h"
|
||||
#include "voe_errors.h"
|
||||
#include "voe_file.h"
|
||||
#include "voe_rtp_rtcp.h"
|
||||
#include "voe_test_defines.h"
|
||||
#include "voe_test_interface.h"
|
||||
|
||||
#include "voe_errors.h"
|
||||
#include "voe_base.h"
|
||||
#include "voe_file.h"
|
||||
#include "voe_dtmf.h"
|
||||
#include "voe_rtp_rtcp.h"
|
||||
#include "voe_audio_processing.h"
|
||||
#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API
|
||||
#include "voe_call_report.h"
|
||||
#endif
|
||||
@ -273,7 +273,11 @@ class VoETestManager {
|
||||
int DoStandardTest();
|
||||
|
||||
const char* AudioFilename() const {
|
||||
return audio_filename_.c_str();
|
||||
const std::string& result = resource_manager_.long_audio_file_path();
|
||||
if (result.length() == 0) {
|
||||
TEST_LOG("ERROR: Failed to open input file!");
|
||||
}
|
||||
return result.c_str();
|
||||
}
|
||||
|
||||
VoiceEngine* VoiceEnginePtr() const {
|
||||
@ -340,7 +344,7 @@ class VoETestManager {
|
||||
int TestNetworkBeforeStreaming();
|
||||
int TestStartStreaming(FakeExternalTransport& channel0_transport);
|
||||
int TestStartPlaying();
|
||||
int TestHoldAndNetEq();
|
||||
int TestNetEq();
|
||||
int TestCodecs();
|
||||
|
||||
bool initialized_;
|
||||
@ -363,8 +367,7 @@ class VoETestManager {
|
||||
VoEVolumeControl* voe_volume_control_;
|
||||
VoEAudioProcessing* voe_apm_;
|
||||
|
||||
std::string resource_path_;
|
||||
std::string audio_filename_;
|
||||
ResourceManager resource_manager_;
|
||||
};
|
||||
|
||||
} // namespace voetest
|
||||
|
@ -22,17 +22,24 @@
|
||||
'include_dirs': [
|
||||
'auto_test',
|
||||
'<(webrtc_root)/modules/interface',
|
||||
# TODO(phoglund): We only depend on voice_engine_defines.h here -
|
||||
# move that file to interface and then remove this dependency.
|
||||
'<(webrtc_root)/voice_engine/main/source',
|
||||
'<(webrtc_root)/modules/audio_device/main/interface',
|
||||
],
|
||||
'sources': [
|
||||
'auto_test/automated_mode.cc',
|
||||
'auto_test/standard/after_initialization_fixture.cc',
|
||||
'auto_test/standard/after_streaming_fixture.cc',
|
||||
'auto_test/standard/before_initialization_fixture.cc',
|
||||
'auto_test/standard/codec_before_streaming_test.cc',
|
||||
'auto_test/standard/hardware_before_initializing_test.cc',
|
||||
'auto_test/standard/hardware_before_streaming_test.cc',
|
||||
'auto_test/standard/manual_hold_test.cc',
|
||||
'auto_test/standard/network_before_streaming_test.cc',
|
||||
'auto_test/standard/rtp_rtcp_before_streaming_test.cc',
|
||||
'auto_test/standard/voe_base_misc_test.cc',
|
||||
'auto_test/resource_manager.cc',
|
||||
'auto_test/voe_cpu_test.cc',
|
||||
'auto_test/voe_cpu_test.h',
|
||||
'auto_test/voe_extended_test.cc',
|
||||
|
Loading…
Reference in New Issue
Block a user